similar to: MFCR2 in Brazil, someone?

Hello there! I've problem with setting up unicall / mfcR2. can't find proper notation for channel, trying unicall/1, unicall/1/1001, unicall/g1, unicall/g1/1000 and still having no luck. klaudia*CLI> !cp call /var/spool/asterisk/outgoing -- Attempting call on Unicall/1001 for application Dial(363) (Retry 1) Mar 25 09:29:34 NOTICE[19920]: channel.c:2429 __ast_request_and_dial:

hi, I am using debian 4.0 with version 2.6.18-4-686 I have downloaded the required files form site asterisk-1.2.24.tar.gz libmfcr2-0.0.3-1.4.tar.bz2 libsupertone-0.0.2.tar.gz libunicall-0.0.3-1.4.tar.bz2 spandsp-20060903.tar.gz I downloaded and installed the files in the follwing sequence spandsp libsupertone libunicall Till here it is compiling and copying

Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia. I have configure my MFCR2 according argentina R2 settigs. (look at the end of the message) the testcall run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2, generally in the begging no errors

Hello I'm using mfcr2 support (unicall) in asterisk 1.4. Everything is working fine, asterisk can answer calls. But after some random period of time mfcr2 module stuck. When I make a call to my * box I can hear only signal of getting caller ID ("tritirirti" - like jumping on :) ) and connection is terminated by my telecom operator. When everything I ok after few seconds of this

Hi, I am having issues sending and receiving fax on my asterisk setup. Currently I have a server that has 2 x E1 TDM cards one is sangoma and the other one is openvox. Both support echo cancellation. One of the e1 is connected to our telco provider via mfcr2 where all our incoming calls originate. On the other end is a pri connection going to HICOM PABX where the local attached to a fax is

Hi List, I'm trying to connect Asterisk with a Nortel Meridian using an E1 with MRFR2 signaling. I've connected both cards, and compiled all the required software... The problem is every call (outgoing or incoming) got dropped, complaining about some "T1 timed out" Only for testing purposes I'm using an application called testcall included on the lib-unicall package, and

Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [006191642208@ps5:1] Dial("SIP/23-081cbc40", "Unicall/g1/91642208|50") in new stack -- Called g1/91642208 [May 14

Is there any way to compile Unicall's libraries (mfcr2, spandsp, chan_unicall, etc) for Asterisk 1.4? BarZ

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi. I am in the Brazil. I have a problem on Unicall for Asterisk 1.4. I am working with E1 with 30 channels. Message when it receives calls: chan_unicall.c:2603 handle_uc_event:unicall/1 protocol error.cause 32772. My files: /etc/zaptel.conf: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone = br defaultzone = br fxoks = 32-55 /etc/asterisk/unicall.conf: [Channels] language=br

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi I have a requirement to bridge Digital ISDN call through an asterisk box but no matter what I setup in the dial plan the second leg of the zap bridge is always set to Transfer Capability of SPEECH, I wondered if any one has come across this and managed to fix it? Thanks in advance for your help Robb

I'm looking to bock a list of numbers users cant call. Is it possible to pull these from file specified in the dial plan, as apposed to mysql?

I need to avoid MOH on my asterisk box, so i need to have a ringing tone when attendant transfer is made, or a call is on hold.. Is there any way to do that. I did not see a simple way to do that. Regards

Hi, I'm wondering what I need to change to get the "swedish" type ring on a SIP-trunk. When I make an inbound call i still have the "US"-type of ring on my SIP trunks. I need help on changing this. However I've successfully changed this on the Zap interface for all inbound calls. Thanks in advance! Regards, Jan

Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop?

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Hi List; I understand that I have to compile zaptel but what about asterisk? Is it enough to extract it? Well, how I will run asterisk (without compilation and installation)? Any advise? Regards Bilal ____________________________________________________________________________________ Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited

Hi all, Are there any issues to be concerned about when calls come in from PSTN to a PRI card and are forwarded back out the same PRI card? Anything different have to be enabled in zaptel.conf or zapata.conf or the Sangoma configs to make this work? What about using .call files that join two ZAP channels? Channel: ZAP/1/4081234567 MaxRetries: 0 RetryTime: 60 WaitTime: 60 Application: